Systems for telephone conferencing over circuit-switched networks, such as traditional public switched telephone networks, are currently in widespread use. Such systems typically use a PSTN audio bridge which allows all participants to a conference call to hear all other participants. Such systems typically generate long-distance telephone fees, and are not able to integrate with video and data with the audio conference.
More recently, systems for enabling audio and/or video conferencing of multiple parties over packet-switched networks, such as the Internet, have become commercially available. Such systems typically allow participants to simultaneously receive and transmit audio and/or video data streams depending on the sophistication of the system. Conferencing systems used over packet-switched networks have the advantage of not generating long-distance telephone fees and enable varying levels of audio, video, and data integration into the conference forum. In a typical system, a conference server receives audio and/or video streams from the participating client processes to the conference, mixes the streams and retransmits the mixed stream to the participating client processes. Such systems provide good audio quality but require considerable dedicated hardware. In such an implementation, each active audio stream from a conference participant must be decompressed, mixed with other audio streams and the mixed audio stream encoded prior to being sent to all participants in the conference. Accordingly, such systems do not scale well due to the costs of the dedicated processing power required for each active audio stream to the conference. A further disadvantage of such systems is that, depending on the compression/decompression (codec) algorithm, much processing power can be expended encoding/decoding and remixing non-active audio data, i.e., silence.
Certain Internet telephone software available today requires participants to a conference to “push” or activate a button on the graphic user interface of the software before speaking, similar to conventional microwave devices such as CB radios. Activation of such a button notifies the conference server that the participant is transmitting an active audio stream. This type of notification scheme is a step backward from current Public Switched Telephone Network conference bridges which allow people to converse in a natural manner.
Still other conference servers transmit only a single audio stream which is considered to be currently active. This technique provides half duplex multi-way audio calling at best. Unfortunately, clipping of the audio stream occurs when the conference server switches between participants, making it difficult to provide a natural multi-way call in which participants may interact, e.g. interrupt each other.
Accordingly, a need exists for a conferencing system in a packet-switched network environment which enables audio and/or video conferencing among participating callers at multiple end points and which is both scalable and does not have cost-prohibitive hardware requirements.
A further need exists for a conferencing system in a packet-switched network environment which enables audio and/or video conferencing among participating callers at multiple end points and which allows participants to interact naturally without requiring activation of the audio stream to speak or which results in audio clipping when switching between participants.
A further need exists for a conferencing system in a packet-switched network environment which enables audio and/or video conferencing among participating callers at multiple end points and which enables the conference server to be implemented in all software.